maxim*CLI> sip debug peer 701 SIP Debugging Enabled for IP: 75.23.69.69:5062 -- Executing Macro("SIP/644-1e76", "exten-vm|novm|701") in new stack -- Executing Macro("SIP/644-1e76", "user-callerid") in new stack -- Executing GotoIf("SIP/644-1e76", "0?report") in new stack -- Executing GotoIf("SIP/644-1e76", "0?start") in new stack -- Executing Set("SIP/644-1e76", "REALCALLERIDNUM=644") in new stack -- Executing NoOp("SIP/644-1e76", "REALCALLERIDNUM is 644") in new stack -- Executing Set("SIP/644-1e76", "AMPUSER=644") in new stack -- Executing Set("SIP/644-1e76", "AMPUSERCIDNAME=Mark Chaney") in new stack -- Executing GotoIf("SIP/644-1e76", "0?report") in new stack -- Executing Set("SIP/644-1e76", "CALLERID(all)=Mark Chaney <644>") in new stack -- Executing NoOp("SIP/644-1e76", "Using CallerID "Mark Chaney" <644>") in new stack -- Executing Set("SIP/644-1e76", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/644-1e76", "VMBOX=novm") in new stack -- Executing Set("SIP/644-1e76", "EXTTOCALL=701") in new stack -- Executing Set("SIP/644-1e76", "CFUEXT=") in new stack -- Executing Set("SIP/644-1e76", "RT=") in new stack -- Executing Macro("SIP/644-1e76", "record-enable|701|IN") in new stack -- Executing GotoIf("SIP/644-1e76", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/644-1e76", "recordingcheck|20061012-170357|1160687037.866") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20061012-170357|1160687037.866: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/644-1e76", "No recording needed") in new stack -- Executing GotoIf("SIP/644-1e76", "0?dolocaldial|1") in new stack -- Executing Macro("SIP/644-1e76", "dial||tr|701") in new stack -- Executing AGI("SIP/644-1e76", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi -- dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'Mark Chaney' number is '644' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 701 to extension map -- dialparties.agi: Extension 701 cf is disabled -- dialparties.agi: Extension 701 do not disturb is disabled > dialparties.agi: extnum: 701 > dialparties.agi: exthascw: 0 > dialparties.agi: exthascfb: 0 > dialparties.agi: extcfb: > dialparties.agi: exthascfu: 0 > dialparties.agi: extcfu: == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 > dialparties.agi: ExtensionState: 0 -- dialparties.agi: Checking CW and CFB status for extension 701 -- dialparties.agi: DbSet CALLTRACE/701 to 644 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("SIP/644-1e76", "SIP/701||tr") in new stack We're at 66.199.224.216 port 14666 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (NAT) to 75.23.69.69:5062: INVITE sip:701@192.168.1.151:5062 SIP/2.0 Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4;rport From: "Mark Chaney" ;tag=as706cc5ae To: Contact: Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 12 Oct 2006 21:03:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 392 392 IN IP4 66.199.224.216 s=session c=IN IP4 66.199.224.216 t=0 0 m=audio 14666 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 701 maxim*CLI> <-- SIP read from 75.23.69.69:5062: SIP/2.0 100 Trying To: From: "Mark Chaney" ;tag=as706cc5ae Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216 CSeq: 102 INVITE Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 --- (8 headers 0 lines)--- maxim*CLI> <-- SIP read from 75.23.69.69:5062: SIP/2.0 180 Ringing To: ;tag=612b8910bf0630d7i0 From: "Mark Chaney" ;tag=as706cc5ae Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216 CSeq: 102 INVITE Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 --- (8 headers 0 lines)--- -- SIP/701-9628 is ringing 12 headers, 0 lines Reliably Transmitting (NAT) to 75.23.69.69:5062: OPTIONS sip:701@192.168.1.151:5062 SIP/2.0 Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK02e580e1;rport From: "Unknown" ;tag=as1f7402e5 To: Contact: Call-ID: 7060e3c902f21f42179febe9594eecde@66.199.224.216 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 12 Oct 2006 21:04:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- maxim*CLI> <-- SIP read from 75.23.69.69:5062: SIP/2.0 200 OK To: ;tag=7a86b35ab04987cdi0 From: "Unknown" ;tag=as1f7402e5 Call-ID: 7060e3c902f21f42179febe9594eecde@66.199.224.216 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK02e580e1 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura --- (10 headers 0 lines)--- Destroying call '7060e3c902f21f42179febe9594eecde@66.199.224.216' Reliably Transmitting (NAT) to 75.23.69.69:5062: CANCEL sip:701@192.168.1.151:5062 SIP/2.0 Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4;rport From: "Mark Chaney" ;tag=as706cc5ae To: Contact: Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '33c3314969e66c014b6522f23e3ffd3a@66.199.224.216' in 32000 ms == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/644-1e76' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/644-1e76' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/644-1e76' maxim*CLI> <-- SIP read from 75.23.69.69:5062: SIP/2.0 487 Request Terminated To: ;tag=612b8910bf0630d7i0 From: "Mark Chaney" ;tag=as706cc5ae Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216 CSeq: 102 INVITE Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 --- (8 headers 0 lines)--- Transmitting (NAT) to 75.23.69.69:5062: ACK sip:701@192.168.1.151:5062 SIP/2.0 Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4;rport From: "Mark Chaney" ;tag=as706cc5ae To: ;tag=612b8910bf0630d7i0 Contact: Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- maxim*CLI> <-- SIP read from 75.23.69.69:5062: SIP/2.0 200 OK To: ;tag=612b8910bf0630d7i0 From: "Mark Chaney" ;tag=as706cc5ae Call-ID: 33c3314969e66c014b6522f23e3ffd3a@66.199.224.216 CSeq: 102 CANCEL Via: SIP/2.0/UDP 66.199.224.216:5060;branch=z9hG4bK3e2c50a4 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '33c3314969e66c014b6522f23e3ffd3a@66.199.224.216' maxim*CLI>